[Asterisk-Users] Mix Data and SIP Phones
I?m looking to install a couple of SIP phones into a small/medium company. The easiest way would be to simply add the phones on the LAN network. But what would happened if someone make a huge file transfer: will it make trouble on the Sip connections ? I think so, that?s why I?m asking you if there is a good (better) way? Maybe connect SIP phones on a separate network and setup a linux box as router w/ traffic shaping ? Thanks. Yves Raber [EMAIL PROTECTED] ___ Dreaming of a Swiss Account? Get it here: http://freemail.swissinfo.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ${CONTEXT}
On Sun, 2004-08-29 at 21:44, Steve Maroney wrote:
I have some problems with my extensions.conf. When a call from pstn comes
in, the call gets put into the [from-fxo] context. From there the caller
is able to dial sip extensions that are included from the [sip-extenions]
context.
When a sip extension is dialed and connected, and then at some point
transfered, the ${CONTEXT} variable is changed from [from-fxo] to
[from-sip]. This leaves the caller from the pstn open to all extenions
that normally only my sip (trusted) clients would be able to dial, such as
outgoing calls on my other FXO ports.
Is the changing on the ${CONTEXT} variable by design (and needs to
secrured in my dialplan) or a bug ?
Post a snippit of your dialplan. Without this, you leave us guessing as
to whether you did the right thing or not.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
On Fri, 27 Aug 2004, Larry Shields wrote: Thanks for the reply. I tried that initially and it did not work. To verify I went back and tried again. It answers and still no sound is heard. -- Accepting call from '8541' to '2688' on channel 0/2, span 1 -- Executing Wait(Zap/2-1, 3) in new stack -- Executing Answer(Zap/2-1, ) in new stack Why do you start with a Wait statement? Just answer the line immediately if you want to do that, or you should at least put a Ringing before the first wait statement if you want the caller to hear a ringing tone before you answer. -- Regards, Tobias Jönsson, Lund SE___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bridging audio in cmd_dial() before connect completes?
On Sun, 29 Aug 2004, Kris Boutilier wrote: Is it possible to make cmd_dial() bridge the audio going out to the network back to the calling party as soon as dial() starts? Put another way, is it possible to have the caller hear the outside dialtone and subsequent DTMF digits? I notice that there is an option 'r' to dial(), thus: [snip] I ask because I'm using EM tie lines from a Norstar, via Asterisk and I get no audio at all after dial() and before the connect status is reached. This may not be of any help to you, but this is how asterisk already works with isdn outgoing lines. The reverse audio path (from the b subscriber to the a subscriber) is opened as soon as the network indicates that the call is proceeding. For analog lines I suspect the audio gets eaten by the channel driver to prevent the dialing party from getting a decond dialtone in their ear. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Revert to dial tone?
On Sun, 29 Aug 2004, Greg Blakely wrote: I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Read the documentation for the Dial command, specifically the 'g' option. Place a DISA command (with no password) as the next priority in the context. Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT dial any digits within a timeout period, + the PBX will return 30 seconds of congestion tone, and then + the PBX will return 60 seconds of howler tone, and then + the extension is 'locked out.' With some timeouts in the context called from the disa application and a chain of contexts (one for each timeout) this should be doable. I'm not sure what you mean by locking out. For a permanent lock you will need to store that information in the AstDB or an external database. Otherwise just Wait(). Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] just-added second X100P
I finally found out the second card was in RED Alarm by running zttool (cat /proc/zaptel/2 also works). The analog extension in our Merlin Legend was bad. Plugging the X100P to another available extension solved the problem. On Sun, 22 Aug 2004 10:53:46 -0500, spectro [EMAIL PROTECTED] wrote: I ran it like 10 times just in case: [EMAIL PROTECTED] asterisk]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Still not working. On Sat, 21 Aug 2004 09:30:19 -0700, Mike Benoit [EMAIL PROTECTED] wrote: Although it shouldn't make a difference, try: channel = 1-2 As well, did you run ztcfg after you installed the new card? I've found sometimes I've had run ztcfg a couple times before Asterisk would kick in and recognize a new card. On Sat, 2004-08-21 at 02:49 -0500, spectro wrote: I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: - fxsks=1-2 loadzone=us defaultzone=us zapata.conf: -- context=inbound-analog signalling=fxs_ks group=1 channel = 1 channel = 2 I created a couple of test extensions: ; test extensions exten = 4390,1,Dial(Zap/g1/4189) exten = 4390,2,Congestion exten = 4391,1,Dial(Zap/1/4189) exten = 4391,2,Congestion exten = 4392,1,Dial(Zap/2/4189) exten = 4392,2,Congestion 4391 works fine, 4392 doesn't: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/2/4189) in new stack Aug 21 02:47:36 NOTICE[426002]: app_dial.c:714 dial_exec: Unable to create chann el of type 'Zap' == Everyone is busy/congested at this time I don't know what's wrong, Zap/2 shows fine in the zap channels list: pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 licenses
[EMAIL PROTECTED] wrote: The G.729 monopolists have made enough money out of their week's work, so why give them more? A better idea is to use a different codec, such as GSM, iLBC or even ulaw (if you have the bandwidth), and ignore G.729 completely. You can add several choices in a list and allow the link to negotiate. This depends highly on what you're doing. If you're on a LAN with plenty of available bandwidth, one of the G.711 variants is the way to go. If you're serving customers on tiny ADSL or cable connections, there's not much else to use than G.729. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100 and call duration
Hi, In the CDR when a call is placed using X100, the saved duration is the one starting with the Zap channel connection, not related to the other part answer. There is any possibility to know when the other part has answered the call placed over X100? I want to know the real call duration in order to be able to compare it with the one provided by my phone company. Thaks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
I've read the manual, for the most part. I've set the volume xml tags all to 1 but when the phone first boots the volume is still not on the highest. The only other thing that might do the trick is the gain options, which I don't understand so that's simply beyond me because I don't want to screw anything up -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, August 30, 2004 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration You need to make the config files yourself, they are all XML and there are sample config files and full descriptions of all fields included with the firmware downloads and admin guides available on the freedomphones.net download link I sent below. There are hundreds of parameters that you can configure to your needs per phone, including speed-dial entries and boot up volume settings. It's worth a read through the admin guide at least once to see what this phone can do. MATT--- -Original Message- From: Matthew Marlowe [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration Atacomm of course :) I looked through the tiki... Couldn't find one understandable configuration file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, August 30, 2004 1:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration Always look in the WIKI: http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones Here's the latest SIP firmware and Admin guides: http://www.freedomphones.net/polycom/files/ Did your IP300 come with the SIP firmware? I've never seen an IP 300 running SIP before, I'd like to know where you got it. Thanks, MATT--- -Original Message- From: Matthew Marlowe [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk + polycom resource and/or willing to help me? Thanks, Im willing to pay something. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI Light On SoundPoint IP 300
I've taken a look at: http://www.voip-info.org/tiki-index.php?page=Getting+MWI+on+Polycom+Phon es+to+work+with+Asterisk I've followed those directions but when I press my message key it doesn't dial the number specified... It actually dials 614p which is the user I register with. My MWI light works, and worked before following those directions, but... I can't make it auto dial the number I tell it to Does anyone have this working? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still unacceptable echo on X101P
I don't have echo problems on my X100P (at home) but that won't stop me from dumping it in favour of a Sipura SPA-3000 next month, once it gets full UK support in firmware (caller ID etc.). It may not be a big deal, but other considerations are: There is another box to manager. Another box to battery backup. Another box to fail. Solving the problem (or atleast determining what causes it) is the more appropriate course of action. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
I think I've found the problem... I don't think my phone is loading the configuration file for some reason. The mac address is correct, it's all lowercase. Everything seems to be set right but I don't think it's reading it. I don't see anything in the log about it. Can someone give some pointers? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Monday, August 30, 2004 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration I've read the manual, for the most part. I've set the volume xml tags all to 1 but when the phone first boots the volume is still not on the highest. The only other thing that might do the trick is the gain options, which I don't understand so that's simply beyond me because I don't want to screw anything up -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, August 30, 2004 2:21 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration You need to make the config files yourself, they are all XML and there are sample config files and full descriptions of all fields included with the firmware downloads and admin guides available on the freedomphones.net download link I sent below. There are hundreds of parameters that you can configure to your needs per phone, including speed-dial entries and boot up volume settings. It's worth a read through the admin guide at least once to see what this phone can do. MATT--- -Original Message- From: Matthew Marlowe [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration Atacomm of course :) I looked through the tiki... Couldn't find one understandable configuration file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, August 30, 2004 1:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration Always look in the WIKI: http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones Here's the latest SIP firmware and Admin guides: http://www.freedomphones.net/polycom/files/ Did your IP300 come with the SIP firmware? I've never seen an IP 300 running SIP before, I'd like to know where you got it. Thanks, MATT--- -Original Message- From: Matthew Marlowe [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk + polycom resource and/or willing to help me? Thanks, Im willing to pay something. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
I'm also interested in this, as the other Steve knows. Anyone in the re-worked jitter-buffer/PLC/DTX crowd besides me going to be at astricon? We can at least start working there on requirements. I think I've wrote this before, but here's what I'd _really_ like to see as requirements for a re-worked jitter-buffer: - Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). - DTX Support: Sending a single CN packet (in IAX2, this should probably be sent reliably) would probably be good. - PLC Support: For codecs that support this natively, use that support (iLBC, speex, others). For those that don't, we can add some kind of hack to the codec. For example, in app_conference, I repeat even GSM frames when I detect loss. - Configurability by applications: It seems that some applications (app_fax) might do better without any of this; we should consider what different applications may want to configure in their use of this, and allow them to change settings while the channel is in the application. So, app_fax might want to disable PLC, and also disable any jitter-buffer shrinkage (and dropped frames that it would cause). - For IAX2 (and RTP, I think) Interleaved frame support: For single-channel calls, in IAX2, and a compressed codec, you have about 100% overhead with 20ms frames. Using larger frame sizes of course drops this considerably, but without interleaving, it makes PLC much less effective. Using interleaved frames, where in a single packet you have, for example Packet 1, frame 0,2 Packet 2, frame 1,3 Packet 3, frame 4,6 [...] And PLC, means that if you drop packet 2, you can do a much better job concealing that loss. -SteveK [EMAIL PROTECTED] wrote: On Fri, 27 Aug 2004, Michael Manousos wrote: I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. Yeah - my goal for a reworked jitter buffer includes DTX and PLC. And other TLAs ;-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How does call routing actually work with SIP?
Daryll Strauss [EMAIL PROTECTED] wrote: On Mon, 2004-08-30 at 09:07, Kevin Walsh wrote: Asterisk will remain in the loop if you have specified t or T in your Dial() command, as it will need to listen for the hash key. It will also remain in the loop if you're recording the audio stream using Monitor(), or whatever. Now if my phone were really smart it would let me reinvite back to Asterisk somehow when I asked to do the transfer, but that would require smarts in the phone/Sipura case which I don't know if that exists. You don't need T or t to transfer most of the time. If you have a SIP phone then it'll probably have a transfer facility anyway. If you're using an analogue phone on an ATA or a Sipura FXS then just use the flash key. By the way, people have been asking about echo. Other than this one test with the busy network I've heard no echos on my Sipura. One thing the Sipura can't do, that I'd like is identify incoming distinctive ring. I have two numbers on the PSTN which differentiate by distinctive ring and I'd like Asterisk to handle them differently. I asked Sipura support and they said they can't do it yet. (Maybe that means a later firmware) Yes - apparently that's due shortly, in a firmware upgrade. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Sayson 480 ADSI
Hello, I have been wanting to try a Sayson 480 ADSI phone with our * box, but I haven't had much luck getting the phone to use *'s built in ADSI script. Does anyone know if there are any how-to's out there for this? Or, could anyone enlighten me? Thanks, Craig Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
- Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete solution... As much as we all hate it's complexity and wish that everything would speak IAX (I know I do) a large number of devices support (and will be supporting) SIP, making it equally as important as IAX2 in using * as a complete telephony solution... DTX Support: Sending a single CN packet (in IAX2, this should probably sent reliably) would probably be good. I second, third and fourth this one as does anyone who's tried to use BroadVoice with Voicemail... Currently when * is not making any noise (e.g. recording) absolutely NO packets are sent back to the proxy... A lot of proxies take this as a sign that the far end has disconnected... Including BroadWorks! But they do recognize small CN packets as a sign that the SIP device (Asterisk) is still there... PLC I think is somewhat implemented already in codecs that support it, but I could be wrong, I remember seeing mention of it in the code... This would be SO helpful!!! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number of simultaneous calls with EM
Hullo over there. i'm trying to link an asterisk box with a legacy PBX system with a four wire trunk line. the legacy PBX has 21 analog phones connected to it and i would like to route calls to another site via the asterisk box. i would like to use EM signaling over this line. my question is how many simultaneous calls can you have over this line with EM signaling. is there a better way of doing this apart from using T1? ___ Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now. http://promotions.yahoo.com/goldrush ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still unacceptable echo on X101P
Brent Franks [EMAIL PROTECTED] wrote: I don't have echo problems on my X100P (at home) but that won't stop me from dumping it in favour of a Sipura SPA-3000 next month, once it gets full UK support in firmware (caller ID etc.). It may not be a big deal, but other considerations are: There is another box to manager. Apart from it being physically in a separate box, I don't see the management problem. You'd configure it once and forget about it, which is what you'd do with a X100P or TDM/FXO; You'd only re-visit the config if you want to tweak something, which is nothing out of the ordinary. Another box to battery backup. You can plug it into the same UPS as your Asterisk server. Again, I don't see the problem. Having each Sipura on a separate UPS would be a bit of an overkill. As they don't draw a lot of power, you could have a bunch of them on a single power strip and plugged into a single socket on the UPS. Another box to fail. A failure would kill one FXO and one FXS. It wouldn't take a server shutdown, and possibly a complete PBX outage, to replace. That's a plus point for the SPA, in my opinion. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom Programmable button Mini Howto and ring state patch
At 1:23 PM -0500 on 8/30/04, David Hinkle wrote: The snom 200 and 220 have five programmable buttons. Each button has a led that can be used to indecate if an extension is idle, in use, or ringing. A button pannel for the 220 is also comming out soon that will have 20'ish programmable buttons on board. This is a killer app for any company that has receptionists handle calls, and pretty usefull for everyone else. As a matter of fact, Asterisk already supports phone idle/in use states for the buttons, and at the bottom of this message you will find a patch to enable the ring state. Howto: 1. Configure the programable buttons as destination and enter the extension in the field. After saving the page the phone will convert the extension to a sip url, which is fine. 2. Modify your asterisk dialplan to provide hints that map a given extension to a channel. (In asterisk, a channel can be busy or ringing, but an extension is just a string of numbers that activate one or more applications). Asterisk seems to provide syntax for allowing more than one channel to be mapped to any particular extension with the hint system, but I did not investigate that. Example: exten = 200,hint,SIP/RonC exten = 200,1,Macro(stdexten,SIP/RonC) exten = 201,hint,SIP/JeanK exten = 201,1,Macro(stdexten,SIP/JeanK) exten = 202,hint,SIP/JeffT exten = 202,1,Macro(stdexten,SIP/JeffT) 3. You must reload the dialplan and then reboot the phone for it's subscriptions to take effect. After that, you should have working lights. 4. If you want the lights to blink on ringing, apply the following patch to the asterisk code. You can not pick up a call by hitting the blinking button, I was going to do this work but I decided to just train the receptionists to hit *8 instead. I have not studied this extensivly, but to implement it, i think it would just require asterisk to have support for sip replaces (I don't know if asterisk supports this or not) and the ringing notify needs to go out with a few more fields. (It seems that the snom phone contacts the sip device listed in one of the ring notify message fields with an invite including a replaces header to pick up a call) I have also included a sip trace of a snom phone picking up a call placed to another phone using the blinking button in case anybody out there wants to tackle this problem themselves (Sample trace was collected when using snom phones with snom's sip proxy software). Please note that it seems like we must include the extra fields in the ring notify before the snom phone will procude the proper replaces invite in order to do a standards compliant call pickup. Notes on patch: If this patch is not in the proper format for submissions please provide me a link to the asterisk submission policies. It has been tested here at DerbyTech for about a week on our live phone system. I submit this patch to the asterisk project under the GPL with hope that it will be resubmited to CVS. Thankyou, David Hinkle Sr. Linux Engineer DerbyTech This is pretty cool! I might get a Snom phone just to try them out. You asked for comments, so here are a few: 1) Send the patch in diff -u format; that's the format used in the bugtracker. 2) You'll need to sign the disclaimer on the http://bugs.digium.com/ interface. This disclaimer doesn't have much of a downside, and all patches to Asterisk for the public CVS have to be disclaimed in this way (avoids SCO-type lawsuits, etc.) 3) Have you looked at the configuration options for the Polycom IP600 phones? I don't know if this trick works with them, but they are pretty slick and have very programmable interfaces which may be almost compatible (or completely compatible) with this method. I haven't looked, but that would be a very cool addition to those phones as well. 4) I'd say you've got 25% of the feature done. Putting the extra effort into having the system pick up the call from any phone when one hits the flashing button would be I think another 25%. Then, the final 50% would be if the button was pressed from a third-party phone while a call was already in progress that all three callers would be bridged together. (more work than it seems, so I give it 50%.) Bit by bit, Asterisk is getting there. Asterisk in general needs to support more PBX-like features. While it says it's an iPBX, it's still falling a bit short when compared to features found in even the most basic key system. See my long posts over time on feature ideas that I've sent to -dev and -users. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Revert to dial tone?
Thanks. That did the trick.
This is what I ended up with (on extension 45)
exten = 45,1,Dial(Zap/44r1,30,g)
exten = 45,2,System(test ${DIALSTATUS} = NOANSWER)
exten = 45,3,GotoIf($[${DIALSTATUS} = NOANSWER]?4:6)
exten = 45,4,voicemail(u10)
exten = 45,5,Hangup
exten = 45,6,DISA(no-password|internal)
exten = 45,7,NoOp
exten = 45,102,voicemail(b10)
exten = 45,103,Hangup
;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Svensson
Sent: Monday, August 30, 2004 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Revert to dial tone?
On Mon, 30 Aug 2004, Greg Blakely wrote:
Thanks. That appears to work, but it doesn't appear to work with
voicemail. From what I can see, the next priority can be taken up
either with the DISA command or the unavailable voicemail command.
Any way of separating the two?
Hm, I guess you want to do different things depending on the
reason for terminating the Dial command? I think there is a
variable DIALSTATUS
that you can test in the dialplan.
Peter
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RE: [Asterisk-Users] G729 licenses
On Mon, 2004-08-30 at 16:26, Kevin Walsh wrote: Brian Wilkins [EMAIL PROTECTED] lazily top-posted: Point is that unfortunately many systems do use G 729 so it is necessary, in order to be compatible with existing gatekeepers, to use that codec. I'd love to use GSM but the existing systems do not support it. It is too costly to re-do everything, therefore you have to work around with what you got. Then you just dump the G.729-only supplier and find one that supports everything else. There are plenty to choose from. I am sorry, but you make it sound a lot simpler than it is. Bottom line is than in reality not everyone has the choice. Without even depending on any supplier(s), I would have issues using gsm, ilbc etc, etc. with our existing equipment. Simple solution dump the $100,000s equipment and find some that does support gsm etc etc. It does not work like that, people have to make commercial decisions and more often than not, that involves compromises. So in it's spirit I would agree with what you say, however in practise ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still unacceptable echo on X101P
On Mon, 30 Aug 2004 10:28:59 -0400, Michael Graves wrote: On Mon, 30 Aug 2004 16:21:04 +0100, Kevin Walsh wrote: Rich Adamson [EMAIL PROTECTED] wrote: 3. If impendance mismatch is the (or a major contributing) factor, can we not devise some interface circuit which will allow a variable rate on the impedance so we can dial out the echo based on individual line conditions? That would be called the TDM card. It has a chipset that was designed to match telco standards in many (if not all) countries. In Canada, if you have x100p echo problems, you'll have them with the tdm card as well. You'll find plenty of discussion in the archives relative to x100p/tdm echo problems unrelated to impedence mismatches. Seems there is an issue with interrupt latency, pci controller, or something like that associated with some motherboards that is impacting echo (as well as other things). Multiple individuals have found that replacing their motherboards fixed the echo problem with these cards, but no one (as yet) as put their finger on why with any degree of accuracy. Lots of opinions, but no real facts to date. [snip] There is a high probability (but this is a wild guess) the echo issue has something to do with the specific chipset used by the motherboard manufacturer, and if that guess is correct, probably something to do with how interrupts are handled or pci controller issues, etc. Something on certain motherboards seem to be delaying the transfer of data to/from x100p/tdm cards, and that delay is sufficient enough to fall outside the limits of the echo cancellation software within *. You could go to a lot of time, effort and expense buying a TDM card, fitting a new motherboard (PCI 2.2), fiddling with gains and trying all of the other suggestions that have been posted, or you could just buy an external FXO device, such as a Sipura SPA-3000, and expect it to just work. External devices are not sensitive to PCI latency and are not concerned with the brand of motherboard you use. All facilities, such as echo cancellation, are provided onboard rather than via a software driver. I don't have echo problems on my X100P (at home) but that won't stop me from dumping it in favour of a Sipura SPA-3000 next month, once it gets full UK support in firmware (caller ID etc.). Bear in mind that many of us who have been using the SPA-3000 in the US have also been experiencing echo related problems. I'm trying the latest firmware which is only a few days old, but I'm pretty close to chucking it in favor of something else. After a day and a half of using the SPA-3000 with v2.1.0 firmware I can say that they echo problem is much better. Not wholly resolved, but better. I still have to jack up the PSTNVOIP gain setting a lot in order to hear the incomming caller. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 The greatest dangers to liberty lurk in insidious encroachments by men of zeal, well-meaning but without understanding. - U.S. Supreme Court Justice Louis Brandeis, 1928 ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXOs
Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Following up on this earlier post we see responses from users with X100/101 cards and Sipura SPA-3000, but nothing else. How about Voicetronix 4 port FXO board? Does anyone have a recommendation or worning about its behaviour with *? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 There ought to be limits to freedom. -- Presidential candidate George W. Bush ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon Reminder: Please register today
Just a brief reminder to everyone who wishes to attend AstriCon 2004: We need your registrations ASAP, especially if you plan on staying on-site at the conference hotel. We have to present the hotel with a solid count of rooms on Wednesday, so please take a few minutes and sign up at: http://www.astricon.net/ As of today we have 189 registrations and the response is growing. We will soon have to cut off registration for the Tutorials due to limited space and materials. If you wish to attend the tutorials you should sign up now. Thanks, Steve Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXOs
Michael Graves wrote: Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Following up on this earlier post we see responses from users with X100/101 cards and Sipura SPA-3000, but nothing else. How about Voicetronix 4 port FXO board? Does anyone have a recommendation or worning about its behaviour with *? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 There ought to be limits to freedom. -- Presidential candidate George W. Bush ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have two TDM400's, one with 4FXO ports and one with 4 FXS ports, and other than a slight tinny sound quality, I have had no problems with them at all. I am not a real expert, but they work fine for me so far. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile error H323
Hi! Have a look at the following entry. I solved this problem: http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error Regards Enrico Stahn smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: does agi wait for digit work in a meetme room ?
From my tests, it doesn't work. - Original Message - From: Eric Bart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 27, 2004 11:12 PM Subject: does agi wait for digit work in a meetme room ? I'd like to monitor key press in a meetme room. Is it possible when connecting one side of a local channel in the meetme room and the other side of the local channel to an agi with the command wait for digit ? Thanks Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch]
This is the message I posted to the asterisk mailing list detailing how
to configure asterisk to drive the snom programmable buttons.
David
---BeginMessage---
The snom 200 and 220 have five programmable buttons. Each button has a
led that can be used to indecate if an extension is idle, in use, or
ringing. A button pannel for the 220 is also comming out soon that will
have 20'ish programmable buttons on board.
This is a killer app for any company that has receptionists handle
calls, and pretty usefull for everyone else.
As a matter of fact, Asterisk already supports phone idle/in use states
for the buttons, and at the bottom of this message you will find a patch
to enable the ring state.
Howto:
1. Configure the programable buttons as destination and enter the
extension in the field. After saving the page the phone will convert
the extension to a sip url, which is fine.
2. Modify your asterisk dialplan to provide hints that map a given
extension to a channel. (In asterisk, a channel can be busy or ringing,
but an extension is just a string of numbers that activate one or more
applications). Asterisk seems to provide syntax for allowing more than
one channel to be mapped to any particular extension with the hint
system, but I did not investigate that.
Example:
exten = 200,hint,SIP/RonC
exten = 200,1,Macro(stdexten,SIP/RonC)
exten = 201,hint,SIP/JeanK
exten = 201,1,Macro(stdexten,SIP/JeanK)
exten = 202,hint,SIP/JeffT
exten = 202,1,Macro(stdexten,SIP/JeffT)
3. You must reload the dialplan and then reboot the phone for it's
subscriptions to take effect. After that, you should have working
lights.
4. If you want the lights to blink on ringing, apply the following
patch to the asterisk code.
You can not pick up a call by hitting the blinking button, I was going
to do this work but I decided to just train the receptionists to hit *8
instead. I have not studied this extensivly, but to implement it, i
think it would just require asterisk to have support for sip replaces
(I don't know if asterisk supports this or not) and the ringing notify
needs to go out with a few more fields. (It seems that the snom phone
contacts the sip device listed in one of the ring notify message fields
with an invite including a replaces header to pick up a call)
I have also included a sip trace of a snom phone picking up a call
placed to another phone using the blinking button in case anybody out
there wants to tackle this problem themselves (Sample trace was
collected when using snom phones with snom's sip proxy software).
Please note that it seems like we must include the extra fields in the
ring notify before the snom phone will procude the proper replaces
invite in order to do a standards compliant call pickup.
Notes on patch:
If this patch is not in the proper format for submissions please provide
me a link to the asterisk submission policies. It has been tested here
at DerbyTech for about a week on our live phone system.
I submit this patch to the asterisk project under the GPL with hope that
it will be resubmited to CVS.
Thankyou,
David Hinkle
Sr. Linux Engineer
DerbyTech
Index: channel.c
===
RCS file: /usr/cvsroot/asterisk/channel.c,v
retrieving revision 1.135
diff -r1.135 channel.c
1910c1910,1915
return AST_DEVICE_INUSE;
---
{
if (chan-_state == AST_STATE_RINGING)
return AST_DEVICE_RINGING;
else
return AST_DEVICE_INUSE;
}
2432a2438
ast_device_state_changed(chan-name);
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.147
diff -r1.147 pbx.c
1371a1372,1373
case AST_DEVICE_RINGING:
return AST_EXTENSION_RINGING;
1377c1379,1381
return AST_EXTENSION_INUSE;
---
//return AST_EXTENSION_INUSE;
allbusy = 0; // Asure we always return INUSE instead of busy because I didn't want to change functionality
// Unless I was ringing
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.483
diff -r1.483 chan_sip.c
3788a3789
char *StateString;
3858a3860,3876
switch(state)
{
case AST_EXTENSION_RINGING:
StateString = early;
break;
case AST_EXTENSION_INUSE:
case AST_EXTENSION_BUSY:
StateString = confirmed;
break;
case AST_EXTENSION_UNAVAILABLE:
case AST_EXTENSION_NOT_INUSE:
default:
StateString = terminated;
}
ast_verbose(State: %s, %d\n, StateString, state);
3870c3888,3889
bytes = snprintf(t, maxbytes, state%s/state\n, state ? confirmed :
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Nevermind, DUH, I was reading it wrong, it states that they DO NOT contain CNG algorithms, it describes a way to send CNG on codecs that do not contain CNG algorithms natively... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPJack
Just a little correction. The link to the company's home page should be http://www.arcturusnetworks.com. But all y'all already figured that out, didn't you? (http://www.arcturus.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delays while playing a message
Hello, 1-2 sec pauses happen while * plays (streams) messages/prompts. We get reports about that from users and experience it ourselves randomly. Cannot reproduce it for debugging though, so need to figure out some other ways to fix it. 1. It's not silence recorded within or pauses between audio files 2. It's not load related - can happen with no load at all 3. We use decent boxes - dual 3.2GHz, 2 Gb RAM, RH9 2.4.22 #4 SMP 2 TE410P (with SMP affinity), PRI 4. It happens on PRI calls 5. Asterisk compiled with ZAPTEL_OPTIMIZATIONS 6. zttest output: --- Results after 84 passes --- Best: 100.00 -- Worst: 99.987793 Questions: 1. Can we ask to explain how buffer underrun is implemented? We could not figure it out by looking at file.c and zaptel with ZAPTEL_OPTIMIZATIONS. 2. Which asterisk parameters/configuration vars could help? Any suggestions on Linux configuration (IO related or any other)? Thank you. Alex Zarubin Webley Systems ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone
now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the hands free mode? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] number of simultaneous calls with EM
I'm not certain I understand your question, however: If you have one four wire EM trunk interface then you can only handle one call over it at a time. EM is a handshaking protocol, not a multiplexing protocol (such as the protocols used by T1 circuits, which give 24 channels on one pair). Hope that helps. Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District -Original Message- From: muhumuza brian [mailto:[EMAIL PROTECTED] Sent: August 30, 2004 1:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] number of simultaneous calls with EM Hullo over there. i'm trying to link an asterisk box with a legacy PBX system with a four wire trunk line. the legacy PBX has 21 analog phones connected to it and i would like to route calls to another site via the asterisk box. i would like to use EM signaling over this line. my question is how many simultaneous calls can you have over this line with EM signaling. is there a better way of doing this apart from using T1? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New to Asterisk and a question
Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until now... I'm fed up with our phone system (or lack thereof) at my office, so I decided to start seriously looking at Asterisk... Mostly as a plaything to get my hands on it, but with the goal of making our phone system here somewhat bearable... We have four incoming POTS lines, which I am going to purchase a TDM400P with four FXO modules to handle... I saw a bundle on Digiums website that I think will suit this requirement nicely... I want to use some Cisco IP phones for each of our desks (there's four of us here)... This is where my question comes in... Does anyone have any recommendations for a model of Cisco IP phone to use? I cannot afford anything expensive -- we're a small company, and to be honest this whole project will be coming out of my personal pocket (I'm part owner of the company, so I can do that and not feel too bad about it. ;) )... Been watching Ebay and saw that Cisco 7940G phones, while expensive, aren't totally out of the question... The next rung down looks ok too (7912G)... Does anyone have any firm objections or praise for these two models? Is there some other make and model that I should look at with similar functionality that may work better with Asterisk? I'm assuming that if the Cisco phones are OK to use, I will have to make them work with SIP? Just looking for opinions on the interoperability of Cisco IP phones and Asterisk. Thanks! Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: New to Asterisk and a question
I recently dug into this, from what I've seen, the best bang for the buck out there is going to be Polycom's. A local vendor has Polycom IP500 phones for $174 shipped to me. IP500 would be comparable to a 7940G I'm assuming. I ran into the same problem with pricing, don't want grandstreams, but can't afford the nice Ciscos. Check out the Polycom's. -Tim -Original Message- From: Brad Stockdale [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 7:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: New to Asterisk and a question Greetings all, I have been watching Asterisk for a while now, but haven't had the nerve to jump in and start playing until now... I'm fed up with our phone system (or lack thereof) at my office, so I decided to start seriously looking at Asterisk... Mostly as a plaything to get my hands on it, but with the goal of making our phone system here somewhat bearable... We have four incoming POTS lines, which I am going to purchase a TDM400P with four FXO modules to handle... I saw a bundle on Digiums website that I think will suit this requirement nicely... I want to use some Cisco IP phones for each of our desks (there's four of us here)... This is where my question comes in... Does anyone have any recommendations for a model of Cisco IP phone to use? I cannot afford anything expensive -- we're a small company, and to be honest this whole project will be coming out of my personal pocket (I'm part owner of the company, so I can do that and not feel too bad about it. ;) )... Been watching Ebay and saw that Cisco 7940G phones, while expensive, aren't totally out of the question... The next rung down looks ok too (7912G)... Does anyone have any firm objections or praise for these two models? Is there some other make and model that I should look at with similar functionality that may work better with Asterisk? I'm assuming that if the Cisco phones are OK to use, I will have to make them work with SIP? Just looking for opinions on the interoperability of Cisco IP phones and Asterisk. Thanks! Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reload crashes Asterisk ?
Hi, I am running Asterisk CVS from 8/27/04, and since about 8/17/04 Asterisk crashes on reload. I did remove support for h323 (as it crashes my * at random, and I don't need it currently). Here is a cut-out of the last lines when I give a reload command... == Parsing '/etc/asterisk/voicemail.conf': Found -- Reloading module 'app_queue.so' (True Call Queueing) == Parsing '/etc/asterisk/queues.conf': Found -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend) -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP)) Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). One funny thing is that when I do an asterisk -rvvvc, I get an old CVS header... == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-HEAD-06/28/04-11:05:26, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-HEAD-06/28/04-11:05:26 currently running on gk2 (pid = 15564) Is this normal ? Warmest Regards, Walter Klomp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Telephony with Asterisk book
It definitely sounded sarcastic :P Lethol On Mon, 30 Aug 2004 08:21:06 -0400, Leif Madsen [EMAIL PROTECTED] wrote: On Mon, 30 Aug 2004 10:21:55 +0800, Joseph Shi [EMAIL PROTECTED] wrote: Steve Underwood Wrote: Just wait for the simplified Chinese version to appear in Shenzhen's Book City. :-) That's great! Will it have the English version as well? Any idea when it will be there? I think he was being sarcastic :) Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceronix and asterisk
Heya Kelvin, Are you using the latest asterisk download from voicetronix webpage. I got most asterisk features working with an OpenLine4 but I still have some bugs/incompatibility issues to resolve. Make sure you download the latest driver and asterisk and make. After installing the voicetronix driver make sure you do the ./echo test included on the README to be sure driver was correctly installed. Lethol On Tue, 31 Aug 2004 00:36:37 +1000, Kelvin And Lisa [EMAIL PROTECTED] wrote: I have installed a 6VPCI card from voicetronix's but i can't get astersik to use it! Now looking at the loaded modules the chan_vpb is not loaded- so I assume that is why it is not working. Now I modified my vpb.conf file and extensions.conf, have I missed something Has anyone a installation guide as I am very new to this!! I have had asterisk working with SIP extensions. by dowloading and making the following Zaptel Libpri asterisk. but after installing the driver for the voicetronix I get errors with the Zaptel when I make it #error modules should never use kernal system header files and the like?? Thanks kelvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenLine4 immediately hangs up on every call
Benjk, I dont have an answer to your problem, but I am currently using the same asterisk CVS HEAD found in voicetronix webpage. Most features are working OK and I am currently trying fo fix a voicemail problem but it appears not to be related to loopdrop. Are you sure the card works fine? (hardware wise) I modded the useloopdrop flag but I have no way of really testing it to see any difference. Make sure you run same context on the vpb.conf and somewhere in your extension.config. II know it sounds newbie-ish, but I am a newbie on asterisk and actually have been finding out the hard way on how to get things working. Anyway, good luck getting it to work. If it does maybe you can try out voicemail from a vpb channel (thats the current problem I am facing) :P Lethol On Mon, 30 Aug 2004 18:33:49 +0900 (JST), Sunrise Ltd [EMAIL PROTECTED] wrote: Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to the same lines. This happens both with UseLoopDrop = 0 and 1 settings in vpb.conf. There don't seem to be any other parameters in the conf file to control this. Has anybody else experienced this? Does anybody know how to teach the vpb driver to behave? BTW, the card is supposed to work in Japan. The console log is provided below... vpb/1-4: chanreads: Got Asterisk bridge with [SIP/2062-70de]. vpb/1-4: chanreads: Checking dtmf's vpb/1-4: chanreads: getting buffer! vpb/1-4: chanreads: got buffer! vpb/1-4: chanreads: applied gain vpb/1-4: chanreads: queueing buffer on read frame q (state[6]) vpb/1-4: Read channel (codec=0) -12 3 vpb/1-4: chanreads: Finished cycle... vpb/1-4: chanreads: Starting cycle ... vpb/1-4: chanreads: Checking bridge vpb/1-4: chanreads: No native bridge. vpb/1-4: chanreads: Got Asterisk bridge with [SIP/2062-70de]. vpb/1-4: chanreads: Checking dtmf's vpb/1-4: chanreads: getting buffer! vpb/1-4: Event [12=[03] Loop Drop] vpb/1-4: Flushing event [12]=[03] Loop Drop vpb/1-4: handle_owned: got event: [12=0] vpb/1-4: handle_owned: putting frame type[4]subclass[1], bridge=(nil) == vpb/1-4: Hangup requested vpb/1-4: chanreads: got buffer! vpb/1-4: chanreads: applied gain vpb/1-4: p-stopreads[1] p-owner[0x8109238] vpb/1-4: chanreads: Finished cycle... == vpb/1-4: Ending record mode (1/yes) vpb/1-4: stopped record thread on vpb/1-4 == vpb/1-4: Ending play mode on vpb/1-4 vpb/1-4: Setting state down == vpb/1-4: Hangup complete Restarting monitor Trying to reawake monitor Monitor restarted == Spawn extension (Internal, 809061554123, 2) exited non-zero on 'SIP/2062-70de' Monitor got null event vpb/1-4: Event [12=[03] Loop Drop] vpb/1-4: Flushing event [12]=[03] Loop Drop vpb/1-4: handle_notowned: mode=3, event[12][[03] Loop Drop ]=[0] vpb/1-4: handle_notowned: mode=3, [12=0] thanks in advance regards benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone
Hmmm... Hands Free might be: voice.gain.rx.digital.chassis=15 (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under ringType John Matthew Marlowe wrote: now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the hands free mode? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Citrix
I have recently played with and did get a working copy of Asterisk functioning in the lab environment. Then when I moved it out of the lab and onto the outside of our firewall it functions as before except that I get a Connected to Asterisk CVS-08/21/04-12:16:04 currently running on 63 (pid = 1478) -- Starting Skinny session from ??.???.114.131 (PIX (6.3) our firewall) Aug 26 14:48:34 WARNING[1209214400]: chan_skinny.c:2275 get_input: Skinny Client sent less data than expected. Aug 26 14:48:34 NOTICE[1209214400]: chan_skinny.c:2335 skinny_session: Skinny Session returned: Success Then once a connection from inside the firewall is made the Citrix Meta Frame fails to connect to the desktop application, if I connect directly to the Meta-frame box outside the desktop application I can connect fine, we found that the port used on the desktop application use port 1604. (Also sometimes referred to as an ICA thin Client) The Citrix Meta frame box sits on the inside of the firewall. So a very simple diagram would be as follows SIP Client XTEN ---Firewall ---(internet) Asterisk | | | | | | Failing Citrix Server - | | | | | | | | ICA Client (metaframe) --- | | | Metaframe Server Has anyone had this issue, or heard of this before Bill Learning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Citrix
chan_skinny refers to support for the cisco Skinny Client Control Protocol which is used by Cisco IP phones in a Cisco Call manager environment. It sounds like the PIX is forwarding stuff to port 2000 on your asterisk box. If you are not using SCCP then you can prevent the module loading by specifying 'noload=skinny.conf' in your asterisk modules.conf Craig - Original Message - From: Learning, Bill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 11:43 AM Subject: [Asterisk-Users] Asterisk and Citrix I have recently played with and did get a working copy of Asterisk functioning in the lab environment. Then when I moved it out of the lab and onto the outside of our firewall it functions as before except that I get a Connected to Asterisk CVS-08/21/04-12:16:04 currently running on 63 (pid = 1478) -- Starting Skinny session from ??.???.114.131 (PIX (6.3) our firewall) Aug 26 14:48:34 WARNING[1209214400]: chan_skinny.c:2275 get_input: Skinny Client sent less data than expected. Aug 26 14:48:34 NOTICE[1209214400]: chan_skinny.c:2335 skinny_session: Skinny Session returned: Success Then once a connection from inside the firewall is made the Citrix Meta Frame fails to connect to the desktop application, if I connect directly to the Meta-frame box outside the desktop application I can connect fine, we found that the port used on the desktop application use port 1604. (Also sometimes referred to as an ICA thin Client) The Citrix Meta frame box sits on the inside of the firewall. So a very simple diagram would be as follows SIP Client XTEN ---Firewall ---(internet) Asterisk | | | | | | Failing Citrix Server - | | | | | | | | ICA Client (metaframe) --- | | | Metaframe Server Has anyone had this issue, or heard of this before Bill Learning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
A caller to my DID's hears Asterisk, but pressing DTMF does nothing:
On call setup iax2 debug shows:
-
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK Timestamp: 2ms SCall: 3 DCall: 00037
[66.234.228.144:4569]
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW Timestamp: 1ms SCall: 00037 DCall: 0
[66.234.228.144:4569]
VERSION : 2
CALLED NUMBER : 5107400469
CALLING NUMBER : 5105408421
CALLING NAME: 5105408421
LANGUAGE: en
CALLED CONTEXT : INGRESS
USERNAME: germanium
FORMAT : 4
CAPABILITY : 4
ADSICPE : 2
DATE TIME : 152969415
CLI iax2 show registry
Host UsernamePerceived Refresh State
66.234.228.170:4569 X 208.184.214.241:4569 60 Registered
---
My config is:
iax.conf:[general]
iax.conf:disallow=all
iax.conf:allow=ulaw
iax.conf:allow=ilbc
iax.conf:allow=gsm
iax.conf:allow=adpcm
iax.conf:allow=alaw
iax.conf:jitterbuffer=no
iax.conf:delayreject=no
iax.conf:register = :[EMAIL PROTECTED]
iax.conf:
iax.conf:[voicepulse-in-01]
iax.conf:type=user
iax.conf:context=voicepulse-test
iax.conf:auth=rsa
iax.conf:inkeys=voicepulse01
extensions.conf:[general]
extensions.conf:static=yes
extensions.conf:writeprotect=yes
extensions.conf:
extensions.conf:[globals]
extensions.conf:[default]
extensions.conf:[voicepulse-test]
extensions.conf:exten = _NXXNXX,1,Playback(beep)
extensions.conf:exten = _NXXNXX,2,SayDigits(${EXTEN})
extensions.conf:exten = _NXXNXX,3,Goto(testdtmf|s|1)
extensions.conf:
extensions.conf:[testdtmf]
extensions.conf:exten = s,1,Background(beep)
extensions.conf:exten = s,2,ResponseTimeout(60)
extensions.conf:exten = _x,1,SayDigits(${EXTEN})
extensions.conf:exten = _x,2,Goto(testdtmf|s|1)
extensions.conf:exten = i,1,Goto(testdtmf|s|1)
extensions.conf:exten = t,1,Hangup
I'm running:
Asterisk CVS-HEAD-08/01/04-22:51:56, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-08/01/04-22:51:56 currently running on
skip (pid = 28611)
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Re: [Asterisk-Users] VoicePulse Connect DTMF with IAX2
Bryce Nesbitt (mailing list account) wrote: Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. I use VoicePulse connect, have similar configs (although I only use iLBC with them) and things are working just fine for me. I just tested with CVS from a day or two ago. I call out and can do DTMF stuff, and likewise if I call in to my DID the caller can navigate my IVRs just fine with DTMF. A data point, I guess. Are you using recent CVS? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My Three-way calls work backwards
I have discovered something that seems to be backwards. When pressing flash to end a three way call Between me, Party A and Party B, Asterisk will drop Party A instead of Party B. My Telcos version of three calling will drop Party B when ending the three way call. In my testing, Party A is a sip client and Party B is a call out through the PSTN. Party A is the first extension i dialed. Party B is the second extension i dialed after press flash to start the three way call. Any ideas ? Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
